Senior Manual QA Engineer (VoIP / Voice / Real-Time Communications)
MMDSmart Ltd
- Казахстан
- Постоянная работа
- Полная занятость
- You will analyze business requirements and technical specifications for both web interfaces and backend, proactively challenging architectural assumptions to prevent issues before they reach production.
- You will act as a "Voice System Investigator" performing deep-dive analysis of complex call flows and media streams using packet captures (PCAPs) and server logs to identify the root cause of signaling and audio defects.
- You will design and execute advanced test strategies that cover the entire communication lifecycle, ensuring seamless integration between WebRTC components, IVR logic, ACD routing, and call recording systems.
- You will investigate the intersection of web application states and real-time media, debugging issues like UI/UX synchronization with agent states, call control race conditions, and WebRTC connectivity hurdles.
- You will proactively simulate network impairments (latency, jitter, packet loss) and use voice quality metrics (MOS) to ensure the system remains resilient and high-performing under sub-optimal conditions.
- You will drive quality improvements within an Agile environment, participating in architectural reviews and brainstorming sessions to advocate for testability and system stability.
- You will mentor the team and improve QA processes, sharing your expertise in troubleshooting stateful, event-driven systems and helping to refine estimation and testing methodologies.
- You will maintain technical documentation and RCA (Root Cause Analysis) reports, ensuring that complex "hard-to-reproduce" bugs are well-documented and that the team learns from every critical leak.
- 5+ years of software testing experience, with at least 2+ years in VoIP / Voice / real-time communication systems
- Strong hands-on understanding of SIP protocol (call flows, registrations, INVITE flows, SDP, RTP)
- Practical experience analyzing SIP/RTP traffic
- Experience testing call center / CCaaS systems (IVR flows; ACD / call routing logic; Call recording; Agent states & call lifecycle)
- Familiarity with WebRTC technologies and real-time communication challenges
- Deep understanding of voice quality metrics (latency, jitter, packet loss, echo, MOS)
- Solid knowledge of TCP/IP networking, NAT, firewalls, ports, STUN/TURN
- Experience working in Linux environments and reading server logs
- Basic scripting skills in Python / JS / TS for test tooling and log analysis
- Experience with REST API testing (Postman, curl, or similar)
- Understanding of RTP media flow vs SIP signaling
- Ability to debug one-way audio, no audio, delayed media, early media
- Experience testing systems that depend on timing, network conditions, and concurrency
- Understanding how telephony behaves differently from web applications (stateful, event-driven, race conditions).
- Experience with test automation using Playwright or similar
- Experience with API automation frameworks (pytest, REST Assured, etc.)
- Familiarity with performance / load tools: (k6, JMeter, Locust)
- Experience simulating SIP traffic using SIPp
- Experience working with PBX / SBC / softswitches (Asterisk, FreeSWITCH, Zoiper, etc.)
- Experience testing under network impairment (packet loss / delay simulation)